similar to: "Called" ID on local extensions

The question below was asked before, and bounced back to me...trying again.... In the Avaya PBX, when a person calls a local extension within the PBX network, the caller is provided with the called person's name on his/her call display (I suppose that this could be called "Called ID"). For example, if I called Joe Blow from my extension 46480 by dialling 41234, I would see "Joe

While testing a problem with "spontaeously" and "occasionally" rebooting asterisk, I came upon this problem: Program received signal SIGSEGV, Segmentation fault. [Switching to Thread -1210770512 (LWP 11346)] 0x002e3fe1 in pri_release_timeout (data=0x93a0478) at q931.c:2589 2589 q931.c: No such file or directory. in q931.c q931.c is in libpri, function

I am using a SIP softphone (X-lite, SIPPS or Firefly) connected to an Asterisk v 1.0.3 PBX. The PBX is also connected via a ISDN-PRI crossover cable to a Avaya Definity Generic 3 PBX via a TE405P card. All outside of the office calls go through the Definity. Here's the issue: Calls to internal SIP extensions, Definity extensions, other offices within our private network (through the

Has anyone been able to successfully integrate the Avaya AUDIX voicemail system with Asterisk? I have been diligently investigating converting our small (Ontario, Canada) office to Asterisk, and ditching our Avaya PBX. However, our head office (New Jersey, USA) maintains our AUDIX system, and a) have no intentions of leaving it and b) some users rely upon AUDIX's ability to transfer

As a follow-up, I have traced the PRI connection and discovered that calls hung up by a local BCM voice station (analog, digital, whatever) send a disconnect with a clear cause of 1 (Number not assigned) instead of a normal clear cause of 16. Calls forwarded through the BCM disconnect with the normal clear cause. I have kludged up a fix to change all clear causes below 2 to 16 (I placed

Guys. Im writing this because Ive checked the wiki, Xten website and read a lot of docs and still cant figure out a way around the NAT issues. Maybe somebody else can give me some ideas from a fresh perpective. My test setup is this: Asterisk -> 2wire homeportal Firewall -> internet Computer with Xten eyebeam The asterisk box and the computer with xten beam are behind the same

-----Original Message----- From: Ferguson, Michael Sent: Tuesday, February 01, 2005 11:35 AM To: 'asterisk-users@lists.digium.com' Subject: Messaging with * and eyeBeam G'Day All, Eyebeam has gotten my interest but I do not have a "high-altitude" view of its interraction with *, therefore my questions. I called xTEN but they preferr to talk to telcos and ISP's

In an attempt to eliminate audio echo I upgraded one side of a working x-lite to x-lite connection to eyebeam. No joy, and what was worse is the audio was even worse now - just noise. Ok, I upgraded the other side to eyebeam and same thing. I'm not even using video (will enable it in sip.conf later, one change at a time). The connection looks something like: eyebeam client

hi, hs anyone able to make video to work on asterisk? i tried following this: http://www.voip-info.org/tiki-index.php?page=Asterisk+phone+xten+eyeBeam i can see that eyebeam is trying to broadcast a video but the other eyebeam is not receiving it. i tested the same setup but this time using ser with rtpproxy and eyebeam video works fine. any ideas? where do you think should i start

G'Day All, Eyebeam has gotten my interest but I do not have a "high-altitude" view of its interraction with *, therefore my questions. I called xTEN but they preferr to talk to telcos and ISP's purchasing hundreds of the eyebeam software... Kind-a stuck here. I already have * happily running and taking care of business on my Windows network. I also have Windows IM running on

Hello I'm using eyeBeam, and Asterisk keeps sending my clients a second call, when they are still in one call (because eyeBeam has lots of channels). I was using X-Lite (with 3 channels) and Asterisk never sent the client a second call. How can I force Asterisk (or eyeBeam) just to send one call at each time. Is this a configuration I need to do in eyeBeam or Asterisk? Thanks Regards Joao

Hi All, So I've been exploring the use of G.722 encoded wideband audio recently. I have three different SIP devices that allow this: Eyebeam, IP650 and a Siemens S865IP. The Siemens and IP650 seems to work fine together. Calls pass between them in what the Polycom notes as "HD" mode and the audio quality is certainly very good. However, things are not so easy with Eyebeam and the

G'Day All, Eyebeam has gotten my interest but I do not have a "high-altitude" view of its interraction with *, therefore my questions. I already have * happily running and taking care of business on my Windows network. I also have Windows IM running on my exchange server and users a IM'ing quite a bit - all internal though-. While I would prefer to use * and eyebeam, I am quite

Hi All, Got some problem with asterisk video, i'm testing an eyebeam and a grandstream video phone. call from grandstream to eyebeam works ok, video shows up. but calls from eyebeam to grandstream there's no video, but audio is ok. Regards, Nhadie

Steve, All, Here is something on that eyeBeam "one end only video support" issue: http://support.xten.net/viewtopic.php?p=15382#15382 Sjobeck ------------------------------ Subject: Re: [Asterisk-Users] Asterisk@home with Eyebeam > I was wondering if its possible to hook up eyebeam with video support to > asterisk@home? Yes. But eyebeam's video support is pretty

Hi all, I am using asterisk1.2.14,realtime and I find there is a strange case in the receiver's display. I have a dial plan to route a call to the destination. I haven't set the callerid(num) for the caller. In the receive ends, it's display shows "asterisk" when I make a call to the receiver. I wonder why "asterisk" shows in the display as I haven't set

I am using eyebeam and I am happy with it. However, it is boring just to talk to my son in the other room. Whenever I try to convince somebody to buy eyebeam, they are scared of the price. Is there a free video soft phone available, that will work with eyebeam / asterisk? bye Ronald Wiplinger

Hello, I have a polycom ip600 and eyebeam. When I call from polycom to eyeBeam, everything, including audio works. When I call the other side (from eyeBeam to polycom), I get no audio. In both cases, eyeBeam shows the same codec: g711u. Also sip show channels shows ulaw codec for both sides and correct addresses. I have canreinvite=no. I don't know if it's important, but asterisk

Hi All. I'm using asterisk 1.2.10, 1.2.13 and 1.4.0-beta3 wth softphone eyebeam 15 I do the following: eyebeam call to PSTN phone 911234567 and asterisk can't create a zap channel sends CANCEL to eyebeam. The log of eyebeam shows this: [06-11-08]16:52:07.415 | Info (debug) RESIP:TRANSACTION | "Matching rule for CANCEL :911234567@192.168.2.54 SIP/2.0 Via: SIP/2.0/UDP

Lets presume that my both software are open. Xlute and Eyebeam But I want my calls from Asterisk to land only on Eyebeam and Not on xlite. How to set it ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090129/4011be6c/attachment.htm